similar to: Remote Voicemail Notifier / enter Dialplan on SIP Register

There is a patch on Mantis (http://bugs.digium.com/view.php?id=4371) Which includes several features. 1. Support for central voicemail server(s) with remote server notification via IAX In other words, this patch allows you to configure an Asterisk server as a central voicemail server and to send out voicemail notification to remote Asterisk servers who can then pass the notification on to

Hello *'s, Every time I started my asterisk error shows: chan_sip.c:Got 200 ok onRegister that isn't a register. Failed to authenticate in Register to <sip:adnan.007@iptel.org>;tag=asfeaa71f Maximum retries exceeded on call xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx@192.168.10.189 for seqno 107 (Request) Registration for adnan.007@195.37.77.99 timed out ,trying again.

The virtio_net driver doesn't dynamically change the RX-ring queue layout and backing pages, but instead reject XDP setup if all the conditions for XDP is not meet. Thus, the xdp_rxq_info also remains fairly static. This allow us to simply add the reg/unreg to net_device open/close functions. Driver hook points for xdp_rxq_info: * reg : virtnet_open * unreg: virtnet_close Cc:

Does Asterisk know when it hits a voicemailbox ? When calling to a cell-phone or GSM, after some rings and no pickup you arrive at a voicemailbox. If Asterisk does not know it's a voicemailbox that has answered the call, the voicemailbox will contain 60minutes of 'silence'. This is very expensive 'silence'. How to avoid this ? Jonas -------------- next part --------------

The virtio_net driver doesn't dynamically change the RX-ring queue layout and backing pages, but instead reject XDP setup if all the conditions for XDP is not meet. Thus, the xdp_rxq_info also remains fairly static. This allow us to simply add the reg/unreg to net_device open/close functions. Driver hook points for xdp_rxq_info: * reg : virtnet_open * unreg: virtnet_close V3: - bugfix,

Hi: when add a new voicemailbox, asterisk will create a new directory to it. since linux has limitation for the number of subdirectory. i wonder how many voicemailbox can asterisk support? thanks. yan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030430/bd36cdaa/attachment.htm

The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox

Hi there, see subject. I'd like to be able to use the vmbox prompt of VoiceMailMain2 and use 1234 and 4321 to point to the same mailbox. Will it be sufficient to create a soft link for 4321 --> 1234 in /var/spool/asterisk/default or will I get myself into horrible trouble? Background: I like to be able to map certain functions ("boss", "peasant",

Anyone else has problems with CVS HEAD's from today with voicemail hanging up silently without any debug/error messages when checked? It also keeps insisting that the user's voice mailbox is full and can't store more messages even if I clear/rebuild the /var/spool/asterisk/voicemail stuff. I've tried falling back to voicemail.conf entries from realtime voicemail with the same

I just completed building a feature rich asterisk voicemail system using perl, php, and mysql. My only concern is that the system i built will not be able to handle the call volume needed. Let me start by explaining my setup. Incoming call -> route.agi (perl -> mysql lookup) -> AGI -> voicemailbox (using mysql odbc) or terminate with wrong number message if a message is left in a

Hi, the app HasNewVoiceMail can't find my voicemail. This is the errormessage: Dec 3 14:24:01 NOTICE[12222481]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 25 at /var/spool/asterisk/voicemail/default/25/(null) does not exist however this is the output of lspbx:~# ls -l /var/spool/asterisk/voicemail/default/25/ total 316 -rwx------ 1 root root 11814 2003-11-22 18:18

The virtio_net driver doesn't dynamically change the RX-ring queue layout and backing pages, but instead reject XDP setup if all the conditions for XDP is not meet. Thus, the xdp_rxq_info also remains fairly static. This allow us to simply add the init+reg/unreg to net_device open/close functions. Driver hook points for xdp_rxq_info: * init+reg: virtnet_open * unreg : virtnet_close Cc:

The virtio_net driver doesn't dynamically change the RX-ring queue layout and backing pages, but instead reject XDP setup if all the conditions for XDP is not meet. Thus, the xdp_rxq_info also remains fairly static. This allow us to simply add the init+reg/unreg to net_device open/close functions. Driver hook points for xdp_rxq_info: * init+reg: virtnet_open * unreg : virtnet_close Cc:

The virtio_net driver doesn't dynamically change the RX-ring queue layout and backing pages, but instead reject XDP setup if all the conditions for XDP is not meet. Thus, the xdp_rxq_info also remains fairly static. This allow us to simply add the reg/unreg to net_device open/close functions. Driver hook points for xdp_rxq_info: * reg : virtnet_open * unreg: virtnet_close V3: - bugfix,

Hi List! I use Asterisk as a pure voicemailbox at a customers place. Right now, a telephone uses up two msns, one for the telephone itself, and one for the telephones mailbox. If the user is absent, a telephonecall is redirected to the voicemail msn of that users telephone. The Problem is: The PBX supports a too small number of msns, so I can't give every user a voicemailbox. Mailboxes are

Hello, I'm having an issue with caller ID in voicemail that I'd appreciate any input on. I have two sip peers defined as extension 100 and 101 each with separate voicemail accounts. Each sip peer has its own DID number, which is established via cid_number = 6021231234. When a call is placed from SIP peer #100 to SIP peer #101, and SIP peer #101 wants to reply to #100's

Hello my friends, I've a doubt, i want to be able to forward the incoming calls from PSTN to my cell phone...i mean, qhen i'm out of the office i need like aq macro that helps me to forward the incoming call that goes for example to my internal extension SIP 207, i 've this macro but i can make it work properly....i can't activate the forward in the phone, is quite confuse:

I'm wondering if there is a simply way to perform a voicemail callback feature using AGI. For instance, a caller leaves a voicemail, the voicemail will then call the owner of the voicemailbox determined by a database look up. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all Looking for a solution to create a flexible voicemail solution in Asterisk without the need to preregister the voicemail users (via databases etc etc). Scenario: All incoming calls are voicemail calls however the dialled number (called party) does not necessarily have a voicemailbox configured in the Asterisk system. I am looking for * to do the following: * Call comes in *

Hi all, When somebody leaves a message in the voicemailbox, is there a way to know the file name of it? I need to return the voicemail file name in the deadagi command. Thanks, Anahi _________________________________________________________________ Convierte las fotos que m?s te gustan en tu nuevo fondo de escritorio para el ordenador. Es f?cil y adem?s gratis